System for suppressing ambient noise in a hands-free device

ABSTRACT

In order to suppress as much noise as possible in a hands-free device in a motor vehicle, for example, two microphones (M 1 , M 2 ) are spaced a certain distance apart, the output signals (MS 1 , MS 2 ) of which are added in an adder (AD) and subtracted in a subtracter (SU). The sum signal (S) of the adder (AD) undergoes a Fourier transform in a first Fourier transformer (F 1 ), and the difference signal (D) of the subtracter (SU) undergoes a Fourier transform in a second Fourier transformer (F 2 ). From the two Fourier transforms R(f) and D(f), a speech pause detector (P) detects speech pauses, during which a third arithmetic unit (R) calculates the transfer function H T  of an adaptive transformation filter (TF). The transfer function of a spectral subtraction filter (SF), at the input of which the Fourier transform R(f) of the sum signal (S) is applied, is generated from the spectral power density S rr  of the sum signal (S) and from the interference power density S nn  generated by the adaptive transformation filter (TF). The output of the spectral subtraction filter (SF) is connected to the input of an inverse Fourier transformer (IF), at the output of which an audio signal (A) can be picked up in the time domain which is essentially free of ambient noise.

1. CLAIM OF PRIORITY

This patent application is a continuation of U.S. patent application Ser. No. 10/497,748 filed Feb. 9, 2005, which is hereby incorporated by reference.

2. FIELD OF THE INVENTION

The invention relates to suppressing ambient noise in a hands-free device having two microphones spaced a predetermined distance apart.

3. RELATED ART

Ambient noise represents a significant interference factor for the use of hands-free devices, which interference factor can significantly degrade the intelligibility of speech. Car phones are equipped with hands-free devices to allow the driver to concentrate fully on driving the vehicle and on traffic. However, particularly loud and interfering ambient noise is encountered in a vehicle.

There is a need for a technique of suppressing ambient noise for a hands-free device.

SUMMARY

A hands-free device is equipped with two microphones spaced a predetermined distance apart. The distance selected for the speaker relative to the microphones is smaller than the so-called diffuse-field distance, so that the direct sound components from the speaker at the location of the microphones predominate over the reflective components occurring within the space.

From the microphone signals supplied by the microphones, the sum and difference signal is generated from which the Fourier transform of the sum signal and the Fourier transform of the difference signal are generated.

From these Fourier transforms, the speech pauses are detected, for example, by determining their average short-term power levels. During speech pauses, the short-term power levels of the sum and difference signal are approximately equal, since for uncorrelated signal components it is unimportant whether these are added or subtracted before the calculation of power, whereas, based on the strongly correlated speech component, when speech begins the short-term power within the sum signal rises significantly relative to the short-term power in the difference signal. This rise is easily detected and exploited to reliably detect a speech pause. As a result, a speech pause can be detected with great reliability even in the case of loud ambient noise.

The spectral power density is determined from the Fourier transform of the sum signal and from the Fourier transform of the difference signal, from which the transfer function for an adaptive transformation filter is calculated. By multiplying the power density of the Fourier transform of the difference signal by its transfer function, this adaptive transformation filter generates the interference power density. From the spectral power density of the Fourier transform of the sum signal and from the interference power density generated by the adaptive transformation filter, the transfer function of an analogous adaptive spectral subtraction filter is calculated that filters the Fourier transform of the sum signal and supplies an audio signal essentially free of ambient noise at its output in the frequency domain, which signal is transformed back to the time domain using an inverse Fourier transform. At the output of this inverse Fourier transform, an audio or speech signal essentially free of ambient noise can be picked up in the time domain and then processed further.

These and other objects, features and advantages of the present invention will become more apparent in light of the following detailed description of preferred embodiments thereof, as illustrated in the accompanying drawing.

DESCRIPTION OF THE DRAWING

The FIGURE is a block diagram illustration of a device for suppressing ambient noise in a hands-free device.

DETAILED DESCRIPTION

The output of a first microphone 100 is provided on a line 102 to an adder 104 and a subtracter 106, while a second microphone 108 provides a sensed signal on a line 110 to the adder 104 and the subtracter 106. The adder 104 provides an output on a line 112 to a first Fourier transformer 114, the output of which on a line 116 is input to a speech pause detector 118, to a first arithmetic unit 120 to calculate the spectral power density S_(rr) of the Fourier transform R(f) of the sum signal, and to an adaptive spectral subtraction filter 122.

The subtracter 106 provides a difference signal on line 124 to a second Fourier transformer 126, the output of which on a line 128 is connected to the speech pause detector 118 and to a second arithmetic unit 130 to calculate the spectral power density S_(DD) of the Fourier transform D(f) of the difference signal on the line 124. The first arithmetic unit 120 provides an output on a line 129 to a third arithmetic unit 132 to calculate the transfer function of an adaptive transformation filter 140, and to the adaptive spectral subtraction filter 122, the output of which is connected to an inverse Fourier transformer 160. The second arithmetic unit 130 provides a signal on line 133, indicative of the spectral power density S_(DD), to the third arithmetic unit 132, and to an adaptive transformation filter 140, the output of which is connected to the adaptive spectral subtraction filter 122. The output of the speech pause detector 118 is also connected to the third arithmetic unit 132, that provides an output which is connected to the control input of the adaptive transformation filter 140.

As mentioned above, the two microphones 100 and 108 are separated a distance which is smaller than the so-called diffuse-field distance. For this reason, the direct sound components of the speaker predominate at the site of the microphone over the reflection components occurring within a closed space, such as the interior of a vehicle.

The short-term power of the Fourier transform R(f) on the line 116 of the sum signal and of the Fourier transform D(f) on the line 128 of the difference signal is determined in the speech pause detector 118. During pauses in speech, the two short-term power levels differ hardly at all since it is unimportant for the uncorrelated speech components whether they are added or subtracted before the power calculation. When speech begins, on the other hand, the short-term power within the sum signal rises significantly relative to the short-term power in the difference signal due to the strongly correlated speech component. This rise thus indicates the end of a speech pause and the beginning of speech.

The first arithmetic unit 120 uses time averaging to calculate the spectral power density S_(rr) of the Fourier transform R(f) on the line 116. Similarly, the second arithmetic unit 130 calculates the spectral power density S_(DD) of the Fourier transform D(f) on the line 128. From the power density S_(rrp)(f) and the spectral power density S_(DDp)(f) during the speech pauses, the third arithmetic unit 132 calculates the transfer function H_(T)(f) of the adaptive transformation filter 140 using the following equation:

H _(T)(f)=S _(rrp)(f)/S _(DDp)(f)  (1)

Preferably, an additional time averaging—that is, a smoothing—of the coefficients of the transfer function thus obtained is used to significantly improve the suppression of ambient noise by preventing the occurrence of so-called artifacts, often called “musical tones.”

The spectral power density S_(rr)(f) is obtained from the Fourier transform R(f) of the sum signal on the line 116 by time averaging, while in analogous fashion the spectral power density S_(DD)(f) is calculated by time averaging from the Fourier transform D(f) of the difference signal on the line 128.

For example, the spectral power density S_(rr) is calculated using the following equation (2):

S _(rr)(f,k)=c*|R(f)|²+(1−c)*S _(rr)(f,k−1)  (2)

In analogous fashion, the spectral power density S_(DD)(f) is, for example, calculated using the equation (3):

S _(DD)(f,k)=c*|D(f)|²+(1−c)*S _(DD)(f,k−1)  (3)

The term c is a constant between 0 and 1 which determines the averaging time period. When c=1, no time averaging takes place; instead the absolute squares of the Fourier transforms R(f) and D(f) are taken as the estimates for the spectral power densities. The calculation of the residual spectral power densities required to implement the method according to the invention is preferably performed in the same manner.

The adaptive transformation filter 140 uses its transfer function H_(T)(f) to generate the interference power density S_(nn) on line 152 from the spectral power density S_(DD)(f) on the line 154 using the following equation (4):

S _(nn)(f)=H _(T) *S _(DD)(f)  (4)

Using the interference power density S_(nn) on the line 152 and the spectral power density S_(rr) on the line 156 the transfer function H_(sub) of the spectral subtraction filter 122 is calculated as specified by equation (5):

H _(sub)(f)=1−a*S _(nn)(f)/S _(rr)(f) for 1−a*S _(nn)(f)/S _(rr)(f)>b

H _(sub)(f)=b for 1−a*S(f)/S _(rr)(f)≦b

The parameter a represents the so-called overestimate factor, while b represents the so-called “spectral floor.”

The interference components picked up by the microphones 100 and 108, which strike the microphones as diffuse sound waves, can be viewed as virtually uncorrelated for almost the entire frequency range of interest. However, there does exist for low frequencies a certain correlation dependent on the relative spacing of the two microphones, which correlation results in the interference components contained in the reference signal appearing to be high-pass-filtered to a certain extent. In order to prevent a faulty estimation of the low-frequency interference components in the spectral subtraction, a spectral boost of the low-frequency components of the reference signal is performed by the adaptive transformation filter 140.

The method according to the invention and the hands-free device according to the invention, which are particularly suitable for a car phone, are distinguished by excellent speech quality and intelligibility since the estimated value for the interference power density S_(nn) on the line 152 is continuously updated independently of the speech activity. As a result, the transfer function of the spectral subtraction filter 122 is also continuously updated, both during speech activity and during speech pauses. As was mentioned above, speech pauses are detected reliably and precisely, this detection being necessary to update the transformation filter 140.

The audio signal at the output on line 158 of the spectral subtraction filter 122, which signal is essentially free of ambient noise, is fed to the inverse Fourier transformer 160 which transforms the audio signal back to the time domain.

Although the present invention has been illustrated and described with respect to several preferred embodiments thereof, various changes, omissions and additions to the form and detail thereof, may be made therein, without departing from the spirit and scope of the invention. 

1. A method of suppressing ambient noise in a hands-free device having two microphones spaced a predetermined distance apart, each of which supplies a microphone signal, comprising: generating a sum signal and a difference signal of the two microphone signals; computing a Fourier transform R(f) of the sum signal (S) and the Fourier transform D(f) of the difference signal (D); detecting speech pauses from the Fourier transforms R(f) and D(f); determining spectral power density S_(rr) from the Fourier transform R(f) of the sum signal (S); determining spectral power density S_(DD) from the Fourier transform D(f) of the difference signal (D); calculating the transfer function H_(T)(f) for an adaptive transformation filter (TF) from the spectral power density S_(rr) of the Fourier transform R(f) of the sum signal (S), and from the spectral power density S_(DD) of the Fourier transform D(f) of the difference signal (D); generating the interference power density S_(nn)(f) by multiplying the power density S_(DD) of the Fourier transform D(f) of the difference signal (D) by its transfer function H_(T)(f); calculating the transfer function H_(sub)(f) of a spectral subtraction filter (SF) from the interference power density S_(nn)(f) and from the spectral power density S_(rr) of the Fourier transform R(f) of the sum signal (S); filtering the Fourier transform R(f) of the sum signal (S) with the spectral subtraction filter (SF); and transforming the output signal of the spectral subtraction filter (SF) back to the time domain.
 2. The method of claim 1, where the transfer function H_(T)(f) of the transformation filter (TF) is generated during speech pauses using the equation: H _(T)(f)=S _(rrp)(f)/S _(DDp)(f)
 3. The method of claim 2, where the coefficients of the transfer function H_(T)(f) of the transformation filter (TF) are averaged over time.
 4. The method of claim 1, where the calculation of the spectral power density S_(rr) from the Fourier transform R(f) of the sum signal (S), and of the spectral power density S_(DD) from the Fourier transform D(f) of the difference signal (D), is performed by time averaging.
 5. The method of claim 4, where the spectral power density S_(rr) is calculated using the equation: S _(rr)(f,k)=c*|R(f)|²+(1−c)*S _(rr)(f,k−1) where k represents the time index, and c is a constant for determining the averaging period.
 6. The method of claim 4, where the spectral power density S_(DD) is calculated using the following equation: S _(DD)(f,k)=c*|D(f)|²+(1−c)*S _(DD)(f,k−1) where k represents a time index, and c is a constant for determining the averaging period.
 7. The method of claim 1, where in order to detect the speech pauses the short-term power of the Fourier transform R(f) of the sum signal (S) and of the Fourier transform D(f) of the difference signal (D) is determined, and that a speech pause is detected whenever the two determined short-term power levels lie within a predetermined common tolerance range.
 8. The method of claim 1, where the transfer function H_(sub)(f) of the spectral subtraction filter (SF) is calculated using the equations: H _(sub)(f)=1−a*S _(nn)(f)/S _(rr)(f) for 1−a*S _(nn)(f)/S _(rr)(f)>b H _(sub)(f)=b for 1−a*S _(nn)(f)/S _(rr)(f)≦b where a represents an overestimation factor and b represents a spectral floor.
 9. The method of claim 1, where the transit time differences between the two microphone signals (MS1, MS2) are equalized.
 10. Hands-free device having two microphones spaced a predetermined distance apart (M1, M2), characterized in that the output of the first microphone (M1) is connected to the first input of an adder (AD) and to the first input of a subtracter (SU); that the output of the second microphone (M2) is connected to the second input of the adder (AD) and the second input of the subtracter (SU); that the output of the adder (AD) is connected to the input of a first Fourier transformer (F1), the output of which is connected to the first input of a speech pause detector (P), to the input of a first arithmetic unit (LS) to calculate the spectral power density S_(rr), and to the input of an adaptive spectral subtraction filter (SF); that the output of the subtracter (SU) is connected to the input of a second Fourier transformer (F2), the output of which is connected to the second input of the speech pause detector (P), and to the input of a second arithmetic unit (LD) to calculate the spectral power density S_(DD); that the outputs of the speech pause detector (P), first arithmetic unit (LS), and second arithmetic unit (LD) are connected to a third arithmetic unit (R) to calculate the transfer function H_(T)(f) of an adaptive transformation filter (TF); that the output of the first arithmetic unit (LS) is connected to the first control input of the adaptive spectral subtraction filter (SF); that the output of the third arithmetic unit (R) is connected to the control input of the adaptive transformation filter (TF), the input of which is connected to the output of the second arithmetic unit (LD), and the output of which is connected to the second control input of the adaptive spectral subtraction filter (SF); and that the output of the adaptive spectral subtraction filter (SF) is connected to the input of an inverse Fourier transformer (IF), at the output of which an audio signal (A) can be picked up which has been transformed back to the time domain.
 11. The hands-free device of claim 10, where the transfer function H_(T)(f) of the transformation filter (TF) is generated during the speech pauses using the following equation: H _(T)(f)=S _(rrp)(f)/S_(DDp)(f)
 12. The hands-free device of claim 11, where the coefficients of the transfer function H_(T)(f) of the transformation filter (TF) are averaged over time.
 13. The hands-free device of claim 10, where the spectral power density S_(rr) is generated by time averaging from the Fourier transform R(f) of the sum signal (S), and that the spectral power density S_(DD) is generated by time averaging from the Fourier transform D(f) of the difference signal (D).
 14. The hands-free device of claim 13, where the spectral power density S_(rr) is generated using the equation: S _(rr)(f,k)=c*|R(f)|²+(1−c)*S _(rr)(f,k−1) where k represents a time index and c is a constant to determine the averaging period.
 15. The hands-free device of claim 13, where the spectral power density S_(DD) is calculated using the equation: S _(DD)(f,k)=c*|D(f)|²+(1−c)*S _(DD)(f,k−1) where k represents a time index, and c is a constant to determine the averaging period.
 16. The hands-free device of claim 10, where the transfer function H_(sub)(f) of the spectral function filter (SF) is calculated using the following equation: H _(sub)(f)=1−a*S _(rr)(f)/S _(rr)(f) for 1−a*S _(nn)(f)/S _(rr)(f)>b H _(sub)(f)=b for 1−a*S _(nn)(f)/S_(rr)(f)≦b where a represents the so-called “overestimate factor” and b represents the “spectral floor.”
 17. The hands-free device of claim 10, where the transit time differences between the two microphone signals (M1, M2) are able to be equalized. 